The following Fedora EPEL 7 Security updates need testing: Age URL 6 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-b169dce5bc chromium-99.0.4844.51-1.el7 4 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-1f3ec359c3 cobbler-2.8.5-5.el7 4 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-db09048bde nbd-3.24-1.el7 2 https://bodhi.fedoraproject.org/updates/FEDORA-EPEL-2022-e1430e72de wordpress-5.1.13-1.el7 The following builds have been pushed to Fedora EPEL 7 updates-testing abcm2ps-8.14.13-1.el7 baresip-2.0.0-1.el7 globus-gssapi-gsi-14.17-4.el7 libre-2.1.1-1.el7 librem-2.0.0-1.el7 zabbix40-4.0.39-1.el7 zabbix50-5.0.21-1.el7 zchunk-1.2.1-1.el7 Details about builds: ================================================================================ abcm2ps-8.14.13-1.el7 (FEDORA-EPEL-2022-d009c17be8) A program to typeset ABC tunes into Postscript -------------------------------------------------------------------------------- Update Information: New upstream bugfix release. -------------------------------------------------------------------------------- ChangeLog: * Sat Mar 12 2022 Stuart Gathman <stuart@xxxxxxxxxxx> - 8.14.13-1 - New upstream release -------------------------------------------------------------------------------- References: [ 1 ] Bug #2063269 - CVE-2021-32434 CVE-2021-32435 CVE-2021-32436 abcm2ps: multiple security vulnerabilities [epel-all] https://bugzilla.redhat.com/show_bug.cgi?id=2063269 -------------------------------------------------------------------------------- ================================================================================ baresip-2.0.0-1.el7 (FEDORA-EPEL-2022-c9b3f4b951) Modular SIP user-agent with audio and video support -------------------------------------------------------------------------------- Update Information: # Baresip 2.0.0 (2022-03-11) - debug_cmd: use `module_event()` for aufileinfo events - multicast: use `module_event()` for sending events - ctrl_dbus: use `module_event()` to send exported event - ua,call: add `CALL_EVENT_OUTGOING` - GTK caller history - Convert FRITZ!Box XML phone book into Baresip contacts - menu: play ringtone on `audio_alert` device - menu: use `str_isset()` for command parameter - dtls_srtp: use elliptic curve cryptography - Support for s16 playback in jack; needed for play tones - Check that account `;sipnat` param has valid value - Tls sipcert per account - Vidsrc add packet handler - ToS for video and sip - account: add accounts parameter to force media address family - Selective early media - ua,uag: split `ua.c` and `uag.c` - Account media af template - account: add missing client certificate parameter to template - account: update answermode values in template - menu: command `uafind` raises UA to head - ctrl_dbus: fix possible memleak on failed initialization - video passthrough - menu: enable auto answer calls also for command dialdir - menu: add command for settings media local direction - Accounts address params - Accounts example cleanup - menu,call: fix hangup for outgoing call - multicast: add source and player API calls - menu: add command `/uareg` - menu: return complete URI for commands `dial`,`dialdir` - menu: in command `dialdir` call `uag_find_requri()` with uri - gst: replace variable length array (buf) with mem_zalloc by @sreimers in #1426 - menu: avoid possible memleaks for `dial`/`dialdir` commands - uag: use local cuser for selecting user-agent - Work on Intercom module - Attended Transfer on GTK - Update `README.md` with configuration suggestion - README fixes - Accounts examples and template - serreg: use a timer for registration restart - gst: audio playback not correct for some WAV files - Working on intercom (ringtone override) - Use line number 0 if user did not provide any line number - AMR Bandwidth Efficient mode support - Working on Intercom (menu: allow other modules to reject a call) - auframe: add samplerate and channels - account: comment out very basic example in template - call answer media dir - Account auto answer beep - serreg: unregister correct User-Agents on registration failure - mk: enable auto-detect of av1 module - ctrl_dbus makefile depends - stream: check if media is present before enabling the RTP timeout - ctrl_dbus: generate dbus code and documentation in makefile - auframe: always set srate and ch - auto answer beep per alert info URI - auframe: move to rem - mixminus: add conference feature - vidbridge: check `vidbridge_disp_display` args fixes segfault - gst: fixed some memory leaks - ua,menu: move auto answer delay handling to menu - ua,menu: move handling of `ANSWERMODE_AUTO` to menu - ausine: support for multiple samplerates by @alfredh in #1479 - account: fix IPv6 only URI for `account_uri_complete()` - ilbc: remove deprecated module - aubridge/device: remove unused `sampv_out` (old resample code) - pkg-config version check - mk: support more locations for `libre.pc` and `librem.pc` - net: remove unused domain - audio: fix `aufilt_setup` update handling - SIP redirect callbackfunction - add secure websocket tls context - test: add `stunuri` - turn: refactoring, add `compv` - fmt: add string to bool function - mk: check glib-2.0 at least like in ubuntu 18.04 - registration fixes - uag,menu: add commands to enable/disable UDP/TCP/TLS - config,audio: add setting `audio.telev_pt` - stream: fix telephone event - Fix I2S compile error, use auframe - ci/tools: fix `pylint` - config: not all audio config was printed - net: replace `network_if_getname` with `net_if_getname` - account: add setting audio payload type for telephone-event - uag,menu: simplify transport enable/disable and support also ws/wss - rst: remove deprecated module - turn: add TCP and TLS transports - speex_pp: remove deprecated module - call: allow video calls by only rejecting a call without any common codecs - multicast: add missing join for multicast addresses - config,uag: rework on `sip_transports` setting - ua: check if peer is capable of video for early video - mqtt/subscribe: replace fixed command buf and increase response size - mqtt: add reconnect handling (lost broker connection) - event: increase `module_event` buffer size - mqtt/subscribe: use safe `odict_string` to prevent crashes - stream: add `stream_set_label` - `Makefile` dependency check improvements - account: add enable/disable flag for video - audio: use account specific audio telev pt correctly - net: add missing `HAVE_INET6` - account: remove unused API function for video enable - gst: changed log level for end of file message - multicast: add new configurable multicast TTL config parameter - call: fix early video capability check (wrong SDP direction checked) - audio: catch end of file message in ausrc error handler - menu: added `stopringing` command - stream: remove obsolete `rx.jbuf_started` - ua: downgrade level of message "ua: using best effort AF" - outgoing calls early callid - audio: changed log level for ausrc error handler messages - SIP default protocol - serreg: fix server selection in case all server were unavailable - multicast: fix missing unlock - config: replace `strcpy` by `saver re_snprintf` - multicast: fix coverity scan - odict: hide struct `odict_entry` - ctrl_dbus: use mqueue to trigger processing of command in remain thread - multicast,config: add separate jitter buffer configuration - ua: emit `CALL_CLOSED` event when user agent is deleted - core: move `stream_enable_rtp_timeout` to api - stream: add mid sdp attribute - rtpext: change length type to `size_t` - avcodec: remove old backwards compat wrapper - main: Added option (`-a`) to set the ua agent string - menu fix tones for parallel outgoing calls - Fix win32 - Fix static analyzer warnings - call: added auto dtmf mode - RTP inbound telephone events should not lead to packet loss - Running tests in a win32 project - stream: wrong media direction after setting stream to hold - move network check to module - serreg: do not ignore returned errors of `ua_register()` - Bundle media mux - mixausrc: no warnings flood when sampc changes - ua: select laddr with route to SDP offer address - net,uag: allow incoming peer-to-peer calls with user@domain - uag: in `uag_reset_transp()` select `laddr` with route to SDP `raddr` - uag: exit if transport could not be added - avcodec: use const AVCodec - module: deprecate module_tmp - test: use ausine as audio source - Selftest fakevideo - When adding local address, check that it has not been added already - start without network - config: add netroam module - multicast: allow any port number for sender and receiver - netroam: add netlink immediate network change detection - remove uag transp rm - net dns srv get - move calls to `stream_start_rtcp` to `call.c` - video: null pointer check for the display handler - audio: add lock - ua: select proper `af` and `laddr` for outgoing IP calls - audio: lock stream - test: replace mock ausrc with ausine - menu ringback session progress - New module providing webrtc aec mobile mode filter - uag: respect setting `sip_listen` - select `laddr` for SDP with respect to `net_interface` - stream: do not start audio during early-video - remove `struct media_ctx` - ci: add libwebrtc-audio-processing-dev (module webrtc_aec) - auconv: new module for audio format conversion - Support for IPv6 link local address for streams - call: check if address family is valid also for video stream - audio: pass pointer to `tx->ausrc_prm` instead of local variable - menu: add an event for call transfer - netroam: error handling for reset transport - mk: use `CC_TEST` for auto detect modules - test: use `dtls_srtp.so` module instead of mock - stream: create `jbuf` only if `use_rtp` is set - multicast: fix memleak in player destructor - stream: split up sender/receiver - set sdp `laddr` to SIP src address - serreg fix fallback accounts - ctrl_dbus: print command with the warning - call: new transfer call state to handle transfered calls correctly - serreg: prevent fast register retries if offline - av1: update packetization code - call: magic check in `sipsess_desc_handler()` - alsa: use `snd_pcm_drop` instead of `snd_pcm_drain` - Increased debian compat level to 10 - conf: fix `conf_configure_buf()` config parse - stream flush rtp socket - Transfer like rfc5589 - GTK: `mem_derefer` call earlier - netroam: add fail counter and event - Added API functions `stream_metric_get_(tx|rx)_bitrate` - Multicast new functions - avcodec: Enable pass-through for more codecs - menu: filter for the correct call state in `menu_selcall` - test: fix warning on mingw32 - menu: Play ringback in play device - sip: add optional TCP source port - rtpext: change id `unsigned` -> `uint8_t` - ci: add mingw build test - test: use mediaenc srtp instead of mock - test: remove mock mediaenc - descr: add `session_description` - use `fs_isfile()` - stream: only call `rtp_clear` for audio - checks if call is available before calling call - conf: add `conf_loadfile` - ice: remove `ice_mode` - audio: use auframe in `encode_rtp_send` - Increased account's max video codec count from four to eight - gtk: Avoid duplicate `call_timer` registration - Attended call transfer by - menu: exclude given call when searching for active call - menu: play call waiting tone on audio_player device - ci/build/macos: link ffmpeg@4 - module auresamp - test: remove h264 testcode, already in retest - h265: move from avcodec to rem - mc: send more details at receiver - timeout event - h265: move packetizer from avcodec to rem - FFmpeg 5 - Fixing clang ThreadSanitizer warnings - auresamp: replace anonymous union for pre C11 compilers - aufile: align naming of alloc handlers - auresamp fixes - mc: new priority handling with multicast state - remove support for Solaris platform - Allow hanging up call that has not been ACKed yet - Multicast identical condition and fmt string fix - audio: allocate aubuf before ausrc_alloc (fixes data race) - call: send supported header for 200 answering/ok - event: check if media line is present for encoding audio/video dir - Removed unused variable in `modules/webrtc_aec/aec.cpp` - audio use module auconv - test: use aufile module - x11grab: remove module, use `avformat.so` instead - audio: declare iterator inside for-loop (C99) - aufile: set `run=true` before write thread starts - Added new API function `call_supported()` and used it in menu module - aufile: separate `aufile_src.c` from `aufile.c` - ctrl_dbus: fix possible data race - menu select other call on hangup - event: encode also combined media direction # libre v2.1.1 (2022-03-12) - mk: fix ABI versioning # libre v2.1.0 (2022-03-11) - Tls sipcert per acc - ToS for video and sip - sdp: in `media_decode()` reset rdir if port is zero - mk/re: add variable length array (`-Wvla`) compiler warning - Macos openssl - `pkg-config` version check - sa: add setter and getter for scope id - net: in `net_dst_source_addr_get()` make parameter `dst` `const` - Avoid `ISO C90 forbids mixed declarations and code` warnings - SIP redirect callbackfunction - add secure websocket tls context - fmt: add string to bool function - fix clang analyze warnings - fmt: support different separators for parameter parsing - Refactor `inet_ntop` and `inet_pton` - add essential fields check - sa: add support for interface suffix for IPv6ll - net: fix `net_if_getname` IPv6 support - udp: add `udp_recv_helper` - sa: fix build for old systems - sa/addrinfo: fix openbsd (drop `AI_V4MAPPED` flag) - ci/codeql: add scan-build - Fixed debian changelog version - IPv6 link local support - sip: add fallback transport for `transp_find()` - SIP default protocol - remove orphaned files - outgoing calls early callid - sip: fix possible "???" dns srv queries by skipping lines without srvid - odict: hide `struct odict_entry` - tls: add keylogger callback function - http/client: support other auth token types besides bearer - tls: fix client certificate replacement - http/client: support dns ipv6 - rtp: add payload-type helper - sip: check consistency between `CSeq` method and that of request line - Fix win32 - fix warnings from PVS-Studio C++ static analyzer - RTP inbound telephone events should not lead to packet loss - support inet6 by default in Win32 project - sdp: differentiate between media line disabled or rejected - move network check to module - odict: move `odict_compare` from retest to re - sip: reuse transport protocol of first request in dialog - json: fix parsing json containing only single value - ice: fix checklist - mk: add `compile_commands.json` (clang only) - sdp: debug print session and media direction - add btrace module (linux/unix only) - mk: add `CC_TEST` header check - init dst address - ice: check if candpair exist before adding - mk: add `CC_TEST` cache - btrace: use `HAVE_EXECINFO` - Coverity - icem: remove dead code (found by coverity 240639) - hash: switch to simpler "fast algorithm" - dns: fix `dnsc_alloc` with IPv6 disabled - mk: deprecate `HAVE_INET6` - Fix for btrace print for memory leaks - set sdp `laddr` to SIP src address - sdp: include all media formats in SDP offer - ci: add centos 7 build test - sip: move `sip_auth_encode` to public api for easier testing - sipsess: do not call desc handler on shutdown - stream flush rtp socket - ci: fix macos openssl build - http: HTTP Host header conform to RFC for IPv6 addresses - Increased debian compatibility level from 9 to 10 - mk: move darwin dns `LFLAGS` to `re.mk` (fixes static builds) - build infrastructure: silent and verbose modes - mk: use posix regex for `sed` `CC` major version detection - dns: fix `parse_resolv_conf` for OpenBSD - sip: add optional TCP source port - ci: add mingw build and test - net: remove `net_hostaddr` - ci/centos7: add openssl - hmac: use `HMAC()` api (fixes OpenSSL 3.0 deprecations) - md5: use `EVP_Digest` for newer openssl versions - sha: add new `sha1()` api - OpenSSL 3.0 - udp: add win32 qos support - ci/mingw: fix dependency checkout - ice: remove `ice_mode` - Codeql security - aubuf insert auframes sorted - ci: add valgrind - tls: remove code for openssl 0.9.5 - ice: remove unused file - main: remove obsolete OPENWRT epoll check - dns,http,sa: fix `HAVE_INET6` off warnings - preliminary support for cmake - make,cmake: set SOVERSION to major version - mk: remove MSVC project files, use cmake instead - natbd: remove module (deprecated) - sha: remove backup implementation - sha,hmac: use Apple CommonCrypto if defined - stun: add `stun_generate_tid` - add cmakelint - Cmake version - cmake: add option to enable/disable rtmp module - lock: use rwlock by default - cmake: fixes for MSVC 16 - json: fix win32 warnings - ci: add cmake build - mqueue: fix win32 warnings - tcp: fix win32 warnings - cmake: fix `target_link_libraries` for win32 - stun: fix win32 warnings - udp: fix win32 warnings - tls: fix win32 warnings - remove `HAVE_INTTYPES_H` - udp: fix win32 warnings - cmake: minor fixes - cmake: fix MSVC ninja - tcp: fix win32 warnings - udp: fix win32 msvc warnings - rtmp: fix win32 warning - bfcp: fix win32 warning - tls: fix libressl 3.5 - fix coverity scan warnings - Allow hanging up call that has not been ACKed yet - mk,cmake: add backtrace support and fix linking on OpenBSD - github: add CMake and Windows workflow - Windows (VS 2022/Ninja) - cmake: fixes for Android - tmr: reuse `tmr_jiffies_usec` - trace: use `gettid` as `thread_id` on linux - tmr: use `CLOCK_MONOTONIC_RAW` if defined - add atomic support - Sonarcloud - sip: fix gcc 6.3.0 warning for logical expression - add transport-cc rtcp feedback support # librem v2.0.0 (2022-03-12) - Restored rgb565 pixel format - vid: remove pixel formats RGB555 and RGB565 - cmake: version 3.7 - mk: bump dev version - au,aulevel: add `AUFMT_S32LE` - aubuf: add `aufbuf_resize()` - cmake: add `HAVE_UNISTD_H` check - cmake: add relative re include dir - cmake: minor fixes - mk: remove win32 project files - cmake: use version 3.10 - aubuf: fix `mem_deref` data race with `frame_destructor ` - h265: move packetizer from avcodec to rem - vidmix: fix `source_put` data race - vidmix: fix possible data race - h265: move `h265_is_keyframe` to rem - h265: move from avcodec to rem - preliminary support for CMake - gitignore: add vim swap and ctags files - ci: fix ccheck main repo path - aubuf: insert audio frames sorted by timestamp - auframe: add `auframe_update` - h264: fix win32 compiler cast warning - mk: bump version v1.0.0-dev3 - Increased debian compatibility level from 9 to 10 - aubuf: remove `aubuf_sort_auframe` return comment - aubuf: add `aubuf_sort_auframe()` - mk: cleanup cache directory - clangd: add config (headers only) - git: ignore clangd files - Fix win32 - mk: bump dev version - aubuf: add auframe functions - add resampler 16<->8 and 32<->16 kHz - aumix: add `aumix_source_mute` - update gitignore for visual studio artifacts - update PlatformToolset to vs2019 - mk: replace `pkg-config` modversion - mk: improve dependency - mk: ignore dependency check on `make clean` - debian: add `pkg- config` file - ci: remove ubuntu-16.04 test - mk: support more locations for `libre.pc` - mk: add `librem.pc` Makefile dependency - mk: add libre version check and pre-release - au/fmt: add `AUFMT_RAW` - auframe: use `enum aufmt` for format - auframe: move from baresip - h264: add functions from baresip - debian: fixes soname pkg build - mk: add abi versioning -------------------------------------------------------------------------------- ChangeLog: * Sun Mar 13 2022 Robert Scheck <robert@xxxxxxxxxxxxxxxxx> 2.0.0-1 - Upgrade to 2.0.0 (#2063451) * Thu Jan 27 2022 Tom Callaway <spot@xxxxxxxxxxxxxxxxx> - 1.1.0-8 - rebuild for libvpx * Wed Jan 19 2022 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 1.1.0-7 - Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild * Sun Dec 5 2021 Richard Shaw <hobbes1069@xxxxxxxxx> - 1.1.0-6 - Rebuild for codec2 1.0.1. -------------------------------------------------------------------------------- References: [ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV https://bugzilla.redhat.com/show_bug.cgi?id=2019879 [ 2 ] Bug #2063340 - libre-2.1.1 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063340 [ 3 ] Bug #2063450 - librem-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063450 [ 4 ] Bug #2063451 - baresip-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063451 -------------------------------------------------------------------------------- ================================================================================ globus-gssapi-gsi-14.17-4.el7 (FEDORA-EPEL-2022-9fd87e9670) Grid Community Toolkit - GSSAPI library -------------------------------------------------------------------------------- Update Information: Fix TLS 1.3 interoperability with dCache gridftp server. -------------------------------------------------------------------------------- ChangeLog: * Sun Mar 6 2022 Mattias Ellert <mattias.ellert@xxxxxxxxxxxxx> - 14.17-4 - Better logic for TLS 1.3 special handling - Use sha256 hash when generating test certificates - Don't test TLS 1.0 and 1.1 when using openssl 3.0.1 or later * Thu Jan 20 2022 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 14.17-3 - Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild * Tue Sep 14 2021 Sahana Prasad <sahana@xxxxxxxxxx> - 14.17-2 - Rebuilt with OpenSSL 3.0.0 -------------------------------------------------------------------------------- ================================================================================ libre-2.1.1-1.el7 (FEDORA-EPEL-2022-c9b3f4b951) Library for real-time communications and SIP stack -------------------------------------------------------------------------------- Update Information: # Baresip 2.0.0 (2022-03-11) - debug_cmd: use `module_event()` for aufileinfo events - multicast: use `module_event()` for sending events - ctrl_dbus: use `module_event()` to send exported event - ua,call: add `CALL_EVENT_OUTGOING` - GTK caller history - Convert FRITZ!Box XML phone book into Baresip contacts - menu: play ringtone on `audio_alert` device - menu: use `str_isset()` for command parameter - dtls_srtp: use elliptic curve cryptography - Support for s16 playback in jack; needed for play tones - Check that account `;sipnat` param has valid value - Tls sipcert per account - Vidsrc add packet handler - ToS for video and sip - account: add accounts parameter to force media address family - Selective early media - ua,uag: split `ua.c` and `uag.c` - Account media af template - account: add missing client certificate parameter to template - account: update answermode values in template - menu: command `uafind` raises UA to head - ctrl_dbus: fix possible memleak on failed initialization - video passthrough - menu: enable auto answer calls also for command dialdir - menu: add command for settings media local direction - Accounts address params - Accounts example cleanup - menu,call: fix hangup for outgoing call - multicast: add source and player API calls - menu: add command `/uareg` - menu: return complete URI for commands `dial`,`dialdir` - menu: in command `dialdir` call `uag_find_requri()` with uri - gst: replace variable length array (buf) with mem_zalloc by @sreimers in #1426 - menu: avoid possible memleaks for `dial`/`dialdir` commands - uag: use local cuser for selecting user-agent - Work on Intercom module - Attended Transfer on GTK - Update `README.md` with configuration suggestion - README fixes - Accounts examples and template - serreg: use a timer for registration restart - gst: audio playback not correct for some WAV files - Working on intercom (ringtone override) - Use line number 0 if user did not provide any line number - AMR Bandwidth Efficient mode support - Working on Intercom (menu: allow other modules to reject a call) - auframe: add samplerate and channels - account: comment out very basic example in template - call answer media dir - Account auto answer beep - serreg: unregister correct User-Agents on registration failure - mk: enable auto-detect of av1 module - ctrl_dbus makefile depends - stream: check if media is present before enabling the RTP timeout - ctrl_dbus: generate dbus code and documentation in makefile - auframe: always set srate and ch - auto answer beep per alert info URI - auframe: move to rem - mixminus: add conference feature - vidbridge: check `vidbridge_disp_display` args fixes segfault - gst: fixed some memory leaks - ua,menu: move auto answer delay handling to menu - ua,menu: move handling of `ANSWERMODE_AUTO` to menu - ausine: support for multiple samplerates by @alfredh in #1479 - account: fix IPv6 only URI for `account_uri_complete()` - ilbc: remove deprecated module - aubridge/device: remove unused `sampv_out` (old resample code) - pkg-config version check - mk: support more locations for `libre.pc` and `librem.pc` - net: remove unused domain - audio: fix `aufilt_setup` update handling - SIP redirect callbackfunction - add secure websocket tls context - test: add `stunuri` - turn: refactoring, add `compv` - fmt: add string to bool function - mk: check glib-2.0 at least like in ubuntu 18.04 - registration fixes - uag,menu: add commands to enable/disable UDP/TCP/TLS - config,audio: add setting `audio.telev_pt` - stream: fix telephone event - Fix I2S compile error, use auframe - ci/tools: fix `pylint` - config: not all audio config was printed - net: replace `network_if_getname` with `net_if_getname` - account: add setting audio payload type for telephone-event - uag,menu: simplify transport enable/disable and support also ws/wss - rst: remove deprecated module - turn: add TCP and TLS transports - speex_pp: remove deprecated module - call: allow video calls by only rejecting a call without any common codecs - multicast: add missing join for multicast addresses - config,uag: rework on `sip_transports` setting - ua: check if peer is capable of video for early video - mqtt/subscribe: replace fixed command buf and increase response size - mqtt: add reconnect handling (lost broker connection) - event: increase `module_event` buffer size - mqtt/subscribe: use safe `odict_string` to prevent crashes - stream: add `stream_set_label` - `Makefile` dependency check improvements - account: add enable/disable flag for video - audio: use account specific audio telev pt correctly - net: add missing `HAVE_INET6` - account: remove unused API function for video enable - gst: changed log level for end of file message - multicast: add new configurable multicast TTL config parameter - call: fix early video capability check (wrong SDP direction checked) - audio: catch end of file message in ausrc error handler - menu: added `stopringing` command - stream: remove obsolete `rx.jbuf_started` - ua: downgrade level of message "ua: using best effort AF" - outgoing calls early callid - audio: changed log level for ausrc error handler messages - SIP default protocol - serreg: fix server selection in case all server were unavailable - multicast: fix missing unlock - config: replace `strcpy` by `saver re_snprintf` - multicast: fix coverity scan - odict: hide struct `odict_entry` - ctrl_dbus: use mqueue to trigger processing of command in remain thread - multicast,config: add separate jitter buffer configuration - ua: emit `CALL_CLOSED` event when user agent is deleted - core: move `stream_enable_rtp_timeout` to api - stream: add mid sdp attribute - rtpext: change length type to `size_t` - avcodec: remove old backwards compat wrapper - main: Added option (`-a`) to set the ua agent string - menu fix tones for parallel outgoing calls - Fix win32 - Fix static analyzer warnings - call: added auto dtmf mode - RTP inbound telephone events should not lead to packet loss - Running tests in a win32 project - stream: wrong media direction after setting stream to hold - move network check to module - serreg: do not ignore returned errors of `ua_register()` - Bundle media mux - mixausrc: no warnings flood when sampc changes - ua: select laddr with route to SDP offer address - net,uag: allow incoming peer-to-peer calls with user@domain - uag: in `uag_reset_transp()` select `laddr` with route to SDP `raddr` - uag: exit if transport could not be added - avcodec: use const AVCodec - module: deprecate module_tmp - test: use ausine as audio source - Selftest fakevideo - When adding local address, check that it has not been added already - start without network - config: add netroam module - multicast: allow any port number for sender and receiver - netroam: add netlink immediate network change detection - remove uag transp rm - net dns srv get - move calls to `stream_start_rtcp` to `call.c` - video: null pointer check for the display handler - audio: add lock - ua: select proper `af` and `laddr` for outgoing IP calls - audio: lock stream - test: replace mock ausrc with ausine - menu ringback session progress - New module providing webrtc aec mobile mode filter - uag: respect setting `sip_listen` - select `laddr` for SDP with respect to `net_interface` - stream: do not start audio during early-video - remove `struct media_ctx` - ci: add libwebrtc-audio-processing-dev (module webrtc_aec) - auconv: new module for audio format conversion - Support for IPv6 link local address for streams - call: check if address family is valid also for video stream - audio: pass pointer to `tx->ausrc_prm` instead of local variable - menu: add an event for call transfer - netroam: error handling for reset transport - mk: use `CC_TEST` for auto detect modules - test: use `dtls_srtp.so` module instead of mock - stream: create `jbuf` only if `use_rtp` is set - multicast: fix memleak in player destructor - stream: split up sender/receiver - set sdp `laddr` to SIP src address - serreg fix fallback accounts - ctrl_dbus: print command with the warning - call: new transfer call state to handle transfered calls correctly - serreg: prevent fast register retries if offline - av1: update packetization code - call: magic check in `sipsess_desc_handler()` - alsa: use `snd_pcm_drop` instead of `snd_pcm_drain` - Increased debian compat level to 10 - conf: fix `conf_configure_buf()` config parse - stream flush rtp socket - Transfer like rfc5589 - GTK: `mem_derefer` call earlier - netroam: add fail counter and event - Added API functions `stream_metric_get_(tx|rx)_bitrate` - Multicast new functions - avcodec: Enable pass-through for more codecs - menu: filter for the correct call state in `menu_selcall` - test: fix warning on mingw32 - menu: Play ringback in play device - sip: add optional TCP source port - rtpext: change id `unsigned` -> `uint8_t` - ci: add mingw build test - test: use mediaenc srtp instead of mock - test: remove mock mediaenc - descr: add `session_description` - use `fs_isfile()` - stream: only call `rtp_clear` for audio - checks if call is available before calling call - conf: add `conf_loadfile` - ice: remove `ice_mode` - audio: use auframe in `encode_rtp_send` - Increased account's max video codec count from four to eight - gtk: Avoid duplicate `call_timer` registration - Attended call transfer by - menu: exclude given call when searching for active call - menu: play call waiting tone on audio_player device - ci/build/macos: link ffmpeg@4 - module auresamp - test: remove h264 testcode, already in retest - h265: move from avcodec to rem - mc: send more details at receiver - timeout event - h265: move packetizer from avcodec to rem - FFmpeg 5 - Fixing clang ThreadSanitizer warnings - auresamp: replace anonymous union for pre C11 compilers - aufile: align naming of alloc handlers - auresamp fixes - mc: new priority handling with multicast state - remove support for Solaris platform - Allow hanging up call that has not been ACKed yet - Multicast identical condition and fmt string fix - audio: allocate aubuf before ausrc_alloc (fixes data race) - call: send supported header for 200 answering/ok - event: check if media line is present for encoding audio/video dir - Removed unused variable in `modules/webrtc_aec/aec.cpp` - audio use module auconv - test: use aufile module - x11grab: remove module, use `avformat.so` instead - audio: declare iterator inside for-loop (C99) - aufile: set `run=true` before write thread starts - Added new API function `call_supported()` and used it in menu module - aufile: separate `aufile_src.c` from `aufile.c` - ctrl_dbus: fix possible data race - menu select other call on hangup - event: encode also combined media direction # libre v2.1.1 (2022-03-12) - mk: fix ABI versioning # libre v2.1.0 (2022-03-11) - Tls sipcert per acc - ToS for video and sip - sdp: in `media_decode()` reset rdir if port is zero - mk/re: add variable length array (`-Wvla`) compiler warning - Macos openssl - `pkg-config` version check - sa: add setter and getter for scope id - net: in `net_dst_source_addr_get()` make parameter `dst` `const` - Avoid `ISO C90 forbids mixed declarations and code` warnings - SIP redirect callbackfunction - add secure websocket tls context - fmt: add string to bool function - fix clang analyze warnings - fmt: support different separators for parameter parsing - Refactor `inet_ntop` and `inet_pton` - add essential fields check - sa: add support for interface suffix for IPv6ll - net: fix `net_if_getname` IPv6 support - udp: add `udp_recv_helper` - sa: fix build for old systems - sa/addrinfo: fix openbsd (drop `AI_V4MAPPED` flag) - ci/codeql: add scan-build - Fixed debian changelog version - IPv6 link local support - sip: add fallback transport for `transp_find()` - SIP default protocol - remove orphaned files - outgoing calls early callid - sip: fix possible "???" dns srv queries by skipping lines without srvid - odict: hide `struct odict_entry` - tls: add keylogger callback function - http/client: support other auth token types besides bearer - tls: fix client certificate replacement - http/client: support dns ipv6 - rtp: add payload-type helper - sip: check consistency between `CSeq` method and that of request line - Fix win32 - fix warnings from PVS-Studio C++ static analyzer - RTP inbound telephone events should not lead to packet loss - support inet6 by default in Win32 project - sdp: differentiate between media line disabled or rejected - move network check to module - odict: move `odict_compare` from retest to re - sip: reuse transport protocol of first request in dialog - json: fix parsing json containing only single value - ice: fix checklist - mk: add `compile_commands.json` (clang only) - sdp: debug print session and media direction - add btrace module (linux/unix only) - mk: add `CC_TEST` header check - init dst address - ice: check if candpair exist before adding - mk: add `CC_TEST` cache - btrace: use `HAVE_EXECINFO` - Coverity - icem: remove dead code (found by coverity 240639) - hash: switch to simpler "fast algorithm" - dns: fix `dnsc_alloc` with IPv6 disabled - mk: deprecate `HAVE_INET6` - Fix for btrace print for memory leaks - set sdp `laddr` to SIP src address - sdp: include all media formats in SDP offer - ci: add centos 7 build test - sip: move `sip_auth_encode` to public api for easier testing - sipsess: do not call desc handler on shutdown - stream flush rtp socket - ci: fix macos openssl build - http: HTTP Host header conform to RFC for IPv6 addresses - Increased debian compatibility level from 9 to 10 - mk: move darwin dns `LFLAGS` to `re.mk` (fixes static builds) - build infrastructure: silent and verbose modes - mk: use posix regex for `sed` `CC` major version detection - dns: fix `parse_resolv_conf` for OpenBSD - sip: add optional TCP source port - ci: add mingw build and test - net: remove `net_hostaddr` - ci/centos7: add openssl - hmac: use `HMAC()` api (fixes OpenSSL 3.0 deprecations) - md5: use `EVP_Digest` for newer openssl versions - sha: add new `sha1()` api - OpenSSL 3.0 - udp: add win32 qos support - ci/mingw: fix dependency checkout - ice: remove `ice_mode` - Codeql security - aubuf insert auframes sorted - ci: add valgrind - tls: remove code for openssl 0.9.5 - ice: remove unused file - main: remove obsolete OPENWRT epoll check - dns,http,sa: fix `HAVE_INET6` off warnings - preliminary support for cmake - make,cmake: set SOVERSION to major version - mk: remove MSVC project files, use cmake instead - natbd: remove module (deprecated) - sha: remove backup implementation - sha,hmac: use Apple CommonCrypto if defined - stun: add `stun_generate_tid` - add cmakelint - Cmake version - cmake: add option to enable/disable rtmp module - lock: use rwlock by default - cmake: fixes for MSVC 16 - json: fix win32 warnings - ci: add cmake build - mqueue: fix win32 warnings - tcp: fix win32 warnings - cmake: fix `target_link_libraries` for win32 - stun: fix win32 warnings - udp: fix win32 warnings - tls: fix win32 warnings - remove `HAVE_INTTYPES_H` - udp: fix win32 warnings - cmake: minor fixes - cmake: fix MSVC ninja - tcp: fix win32 warnings - udp: fix win32 msvc warnings - rtmp: fix win32 warning - bfcp: fix win32 warning - tls: fix libressl 3.5 - fix coverity scan warnings - Allow hanging up call that has not been ACKed yet - mk,cmake: add backtrace support and fix linking on OpenBSD - github: add CMake and Windows workflow - Windows (VS 2022/Ninja) - cmake: fixes for Android - tmr: reuse `tmr_jiffies_usec` - trace: use `gettid` as `thread_id` on linux - tmr: use `CLOCK_MONOTONIC_RAW` if defined - add atomic support - Sonarcloud - sip: fix gcc 6.3.0 warning for logical expression - add transport-cc rtcp feedback support # librem v2.0.0 (2022-03-12) - Restored rgb565 pixel format - vid: remove pixel formats RGB555 and RGB565 - cmake: version 3.7 - mk: bump dev version - au,aulevel: add `AUFMT_S32LE` - aubuf: add `aufbuf_resize()` - cmake: add `HAVE_UNISTD_H` check - cmake: add relative re include dir - cmake: minor fixes - mk: remove win32 project files - cmake: use version 3.10 - aubuf: fix `mem_deref` data race with `frame_destructor ` - h265: move packetizer from avcodec to rem - vidmix: fix `source_put` data race - vidmix: fix possible data race - h265: move `h265_is_keyframe` to rem - h265: move from avcodec to rem - preliminary support for CMake - gitignore: add vim swap and ctags files - ci: fix ccheck main repo path - aubuf: insert audio frames sorted by timestamp - auframe: add `auframe_update` - h264: fix win32 compiler cast warning - mk: bump version v1.0.0-dev3 - Increased debian compatibility level from 9 to 10 - aubuf: remove `aubuf_sort_auframe` return comment - aubuf: add `aubuf_sort_auframe()` - mk: cleanup cache directory - clangd: add config (headers only) - git: ignore clangd files - Fix win32 - mk: bump dev version - aubuf: add auframe functions - add resampler 16<->8 and 32<->16 kHz - aumix: add `aumix_source_mute` - update gitignore for visual studio artifacts - update PlatformToolset to vs2019 - mk: replace `pkg-config` modversion - mk: improve dependency - mk: ignore dependency check on `make clean` - debian: add `pkg- config` file - ci: remove ubuntu-16.04 test - mk: support more locations for `libre.pc` - mk: add `librem.pc` Makefile dependency - mk: add libre version check and pre-release - au/fmt: add `AUFMT_RAW` - auframe: use `enum aufmt` for format - auframe: move from baresip - h264: add functions from baresip - debian: fixes soname pkg build - mk: add abi versioning -------------------------------------------------------------------------------- ChangeLog: * Sun Mar 13 2022 Robert Scheck <robert@xxxxxxxxxxxxxxxxx> 2.1.1-1 - Upgrade to 2.1.1 (#2063340) * Fri Mar 11 2022 Robert Scheck <robert@xxxxxxxxxxxxxxxxx> 2.1.0-1 - Upgrade to 2.1.0 (#2063340) * Thu Jan 20 2022 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 2.0.1-4 - Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild * Tue Sep 14 2021 Sahana Prasad <sahana@xxxxxxxxxx> - 2.0.1-3 - Rebuilt with OpenSSL 3.0.0 * Thu Jul 22 2021 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 2.0.1-2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild -------------------------------------------------------------------------------- References: [ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV https://bugzilla.redhat.com/show_bug.cgi?id=2019879 [ 2 ] Bug #2063340 - libre-2.1.1 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063340 [ 3 ] Bug #2063450 - librem-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063450 [ 4 ] Bug #2063451 - baresip-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063451 -------------------------------------------------------------------------------- ================================================================================ librem-2.0.0-1.el7 (FEDORA-EPEL-2022-c9b3f4b951) Library for real-time audio and video processing -------------------------------------------------------------------------------- Update Information: # Baresip 2.0.0 (2022-03-11) - debug_cmd: use `module_event()` for aufileinfo events - multicast: use `module_event()` for sending events - ctrl_dbus: use `module_event()` to send exported event - ua,call: add `CALL_EVENT_OUTGOING` - GTK caller history - Convert FRITZ!Box XML phone book into Baresip contacts - menu: play ringtone on `audio_alert` device - menu: use `str_isset()` for command parameter - dtls_srtp: use elliptic curve cryptography - Support for s16 playback in jack; needed for play tones - Check that account `;sipnat` param has valid value - Tls sipcert per account - Vidsrc add packet handler - ToS for video and sip - account: add accounts parameter to force media address family - Selective early media - ua,uag: split `ua.c` and `uag.c` - Account media af template - account: add missing client certificate parameter to template - account: update answermode values in template - menu: command `uafind` raises UA to head - ctrl_dbus: fix possible memleak on failed initialization - video passthrough - menu: enable auto answer calls also for command dialdir - menu: add command for settings media local direction - Accounts address params - Accounts example cleanup - menu,call: fix hangup for outgoing call - multicast: add source and player API calls - menu: add command `/uareg` - menu: return complete URI for commands `dial`,`dialdir` - menu: in command `dialdir` call `uag_find_requri()` with uri - gst: replace variable length array (buf) with mem_zalloc by @sreimers in #1426 - menu: avoid possible memleaks for `dial`/`dialdir` commands - uag: use local cuser for selecting user-agent - Work on Intercom module - Attended Transfer on GTK - Update `README.md` with configuration suggestion - README fixes - Accounts examples and template - serreg: use a timer for registration restart - gst: audio playback not correct for some WAV files - Working on intercom (ringtone override) - Use line number 0 if user did not provide any line number - AMR Bandwidth Efficient mode support - Working on Intercom (menu: allow other modules to reject a call) - auframe: add samplerate and channels - account: comment out very basic example in template - call answer media dir - Account auto answer beep - serreg: unregister correct User-Agents on registration failure - mk: enable auto-detect of av1 module - ctrl_dbus makefile depends - stream: check if media is present before enabling the RTP timeout - ctrl_dbus: generate dbus code and documentation in makefile - auframe: always set srate and ch - auto answer beep per alert info URI - auframe: move to rem - mixminus: add conference feature - vidbridge: check `vidbridge_disp_display` args fixes segfault - gst: fixed some memory leaks - ua,menu: move auto answer delay handling to menu - ua,menu: move handling of `ANSWERMODE_AUTO` to menu - ausine: support for multiple samplerates by @alfredh in #1479 - account: fix IPv6 only URI for `account_uri_complete()` - ilbc: remove deprecated module - aubridge/device: remove unused `sampv_out` (old resample code) - pkg-config version check - mk: support more locations for `libre.pc` and `librem.pc` - net: remove unused domain - audio: fix `aufilt_setup` update handling - SIP redirect callbackfunction - add secure websocket tls context - test: add `stunuri` - turn: refactoring, add `compv` - fmt: add string to bool function - mk: check glib-2.0 at least like in ubuntu 18.04 - registration fixes - uag,menu: add commands to enable/disable UDP/TCP/TLS - config,audio: add setting `audio.telev_pt` - stream: fix telephone event - Fix I2S compile error, use auframe - ci/tools: fix `pylint` - config: not all audio config was printed - net: replace `network_if_getname` with `net_if_getname` - account: add setting audio payload type for telephone-event - uag,menu: simplify transport enable/disable and support also ws/wss - rst: remove deprecated module - turn: add TCP and TLS transports - speex_pp: remove deprecated module - call: allow video calls by only rejecting a call without any common codecs - multicast: add missing join for multicast addresses - config,uag: rework on `sip_transports` setting - ua: check if peer is capable of video for early video - mqtt/subscribe: replace fixed command buf and increase response size - mqtt: add reconnect handling (lost broker connection) - event: increase `module_event` buffer size - mqtt/subscribe: use safe `odict_string` to prevent crashes - stream: add `stream_set_label` - `Makefile` dependency check improvements - account: add enable/disable flag for video - audio: use account specific audio telev pt correctly - net: add missing `HAVE_INET6` - account: remove unused API function for video enable - gst: changed log level for end of file message - multicast: add new configurable multicast TTL config parameter - call: fix early video capability check (wrong SDP direction checked) - audio: catch end of file message in ausrc error handler - menu: added `stopringing` command - stream: remove obsolete `rx.jbuf_started` - ua: downgrade level of message "ua: using best effort AF" - outgoing calls early callid - audio: changed log level for ausrc error handler messages - SIP default protocol - serreg: fix server selection in case all server were unavailable - multicast: fix missing unlock - config: replace `strcpy` by `saver re_snprintf` - multicast: fix coverity scan - odict: hide struct `odict_entry` - ctrl_dbus: use mqueue to trigger processing of command in remain thread - multicast,config: add separate jitter buffer configuration - ua: emit `CALL_CLOSED` event when user agent is deleted - core: move `stream_enable_rtp_timeout` to api - stream: add mid sdp attribute - rtpext: change length type to `size_t` - avcodec: remove old backwards compat wrapper - main: Added option (`-a`) to set the ua agent string - menu fix tones for parallel outgoing calls - Fix win32 - Fix static analyzer warnings - call: added auto dtmf mode - RTP inbound telephone events should not lead to packet loss - Running tests in a win32 project - stream: wrong media direction after setting stream to hold - move network check to module - serreg: do not ignore returned errors of `ua_register()` - Bundle media mux - mixausrc: no warnings flood when sampc changes - ua: select laddr with route to SDP offer address - net,uag: allow incoming peer-to-peer calls with user@domain - uag: in `uag_reset_transp()` select `laddr` with route to SDP `raddr` - uag: exit if transport could not be added - avcodec: use const AVCodec - module: deprecate module_tmp - test: use ausine as audio source - Selftest fakevideo - When adding local address, check that it has not been added already - start without network - config: add netroam module - multicast: allow any port number for sender and receiver - netroam: add netlink immediate network change detection - remove uag transp rm - net dns srv get - move calls to `stream_start_rtcp` to `call.c` - video: null pointer check for the display handler - audio: add lock - ua: select proper `af` and `laddr` for outgoing IP calls - audio: lock stream - test: replace mock ausrc with ausine - menu ringback session progress - New module providing webrtc aec mobile mode filter - uag: respect setting `sip_listen` - select `laddr` for SDP with respect to `net_interface` - stream: do not start audio during early-video - remove `struct media_ctx` - ci: add libwebrtc-audio-processing-dev (module webrtc_aec) - auconv: new module for audio format conversion - Support for IPv6 link local address for streams - call: check if address family is valid also for video stream - audio: pass pointer to `tx->ausrc_prm` instead of local variable - menu: add an event for call transfer - netroam: error handling for reset transport - mk: use `CC_TEST` for auto detect modules - test: use `dtls_srtp.so` module instead of mock - stream: create `jbuf` only if `use_rtp` is set - multicast: fix memleak in player destructor - stream: split up sender/receiver - set sdp `laddr` to SIP src address - serreg fix fallback accounts - ctrl_dbus: print command with the warning - call: new transfer call state to handle transfered calls correctly - serreg: prevent fast register retries if offline - av1: update packetization code - call: magic check in `sipsess_desc_handler()` - alsa: use `snd_pcm_drop` instead of `snd_pcm_drain` - Increased debian compat level to 10 - conf: fix `conf_configure_buf()` config parse - stream flush rtp socket - Transfer like rfc5589 - GTK: `mem_derefer` call earlier - netroam: add fail counter and event - Added API functions `stream_metric_get_(tx|rx)_bitrate` - Multicast new functions - avcodec: Enable pass-through for more codecs - menu: filter for the correct call state in `menu_selcall` - test: fix warning on mingw32 - menu: Play ringback in play device - sip: add optional TCP source port - rtpext: change id `unsigned` -> `uint8_t` - ci: add mingw build test - test: use mediaenc srtp instead of mock - test: remove mock mediaenc - descr: add `session_description` - use `fs_isfile()` - stream: only call `rtp_clear` for audio - checks if call is available before calling call - conf: add `conf_loadfile` - ice: remove `ice_mode` - audio: use auframe in `encode_rtp_send` - Increased account's max video codec count from four to eight - gtk: Avoid duplicate `call_timer` registration - Attended call transfer by - menu: exclude given call when searching for active call - menu: play call waiting tone on audio_player device - ci/build/macos: link ffmpeg@4 - module auresamp - test: remove h264 testcode, already in retest - h265: move from avcodec to rem - mc: send more details at receiver - timeout event - h265: move packetizer from avcodec to rem - FFmpeg 5 - Fixing clang ThreadSanitizer warnings - auresamp: replace anonymous union for pre C11 compilers - aufile: align naming of alloc handlers - auresamp fixes - mc: new priority handling with multicast state - remove support for Solaris platform - Allow hanging up call that has not been ACKed yet - Multicast identical condition and fmt string fix - audio: allocate aubuf before ausrc_alloc (fixes data race) - call: send supported header for 200 answering/ok - event: check if media line is present for encoding audio/video dir - Removed unused variable in `modules/webrtc_aec/aec.cpp` - audio use module auconv - test: use aufile module - x11grab: remove module, use `avformat.so` instead - audio: declare iterator inside for-loop (C99) - aufile: set `run=true` before write thread starts - Added new API function `call_supported()` and used it in menu module - aufile: separate `aufile_src.c` from `aufile.c` - ctrl_dbus: fix possible data race - menu select other call on hangup - event: encode also combined media direction # libre v2.1.1 (2022-03-12) - mk: fix ABI versioning # libre v2.1.0 (2022-03-11) - Tls sipcert per acc - ToS for video and sip - sdp: in `media_decode()` reset rdir if port is zero - mk/re: add variable length array (`-Wvla`) compiler warning - Macos openssl - `pkg-config` version check - sa: add setter and getter for scope id - net: in `net_dst_source_addr_get()` make parameter `dst` `const` - Avoid `ISO C90 forbids mixed declarations and code` warnings - SIP redirect callbackfunction - add secure websocket tls context - fmt: add string to bool function - fix clang analyze warnings - fmt: support different separators for parameter parsing - Refactor `inet_ntop` and `inet_pton` - add essential fields check - sa: add support for interface suffix for IPv6ll - net: fix `net_if_getname` IPv6 support - udp: add `udp_recv_helper` - sa: fix build for old systems - sa/addrinfo: fix openbsd (drop `AI_V4MAPPED` flag) - ci/codeql: add scan-build - Fixed debian changelog version - IPv6 link local support - sip: add fallback transport for `transp_find()` - SIP default protocol - remove orphaned files - outgoing calls early callid - sip: fix possible "???" dns srv queries by skipping lines without srvid - odict: hide `struct odict_entry` - tls: add keylogger callback function - http/client: support other auth token types besides bearer - tls: fix client certificate replacement - http/client: support dns ipv6 - rtp: add payload-type helper - sip: check consistency between `CSeq` method and that of request line - Fix win32 - fix warnings from PVS-Studio C++ static analyzer - RTP inbound telephone events should not lead to packet loss - support inet6 by default in Win32 project - sdp: differentiate between media line disabled or rejected - move network check to module - odict: move `odict_compare` from retest to re - sip: reuse transport protocol of first request in dialog - json: fix parsing json containing only single value - ice: fix checklist - mk: add `compile_commands.json` (clang only) - sdp: debug print session and media direction - add btrace module (linux/unix only) - mk: add `CC_TEST` header check - init dst address - ice: check if candpair exist before adding - mk: add `CC_TEST` cache - btrace: use `HAVE_EXECINFO` - Coverity - icem: remove dead code (found by coverity 240639) - hash: switch to simpler "fast algorithm" - dns: fix `dnsc_alloc` with IPv6 disabled - mk: deprecate `HAVE_INET6` - Fix for btrace print for memory leaks - set sdp `laddr` to SIP src address - sdp: include all media formats in SDP offer - ci: add centos 7 build test - sip: move `sip_auth_encode` to public api for easier testing - sipsess: do not call desc handler on shutdown - stream flush rtp socket - ci: fix macos openssl build - http: HTTP Host header conform to RFC for IPv6 addresses - Increased debian compatibility level from 9 to 10 - mk: move darwin dns `LFLAGS` to `re.mk` (fixes static builds) - build infrastructure: silent and verbose modes - mk: use posix regex for `sed` `CC` major version detection - dns: fix `parse_resolv_conf` for OpenBSD - sip: add optional TCP source port - ci: add mingw build and test - net: remove `net_hostaddr` - ci/centos7: add openssl - hmac: use `HMAC()` api (fixes OpenSSL 3.0 deprecations) - md5: use `EVP_Digest` for newer openssl versions - sha: add new `sha1()` api - OpenSSL 3.0 - udp: add win32 qos support - ci/mingw: fix dependency checkout - ice: remove `ice_mode` - Codeql security - aubuf insert auframes sorted - ci: add valgrind - tls: remove code for openssl 0.9.5 - ice: remove unused file - main: remove obsolete OPENWRT epoll check - dns,http,sa: fix `HAVE_INET6` off warnings - preliminary support for cmake - make,cmake: set SOVERSION to major version - mk: remove MSVC project files, use cmake instead - natbd: remove module (deprecated) - sha: remove backup implementation - sha,hmac: use Apple CommonCrypto if defined - stun: add `stun_generate_tid` - add cmakelint - Cmake version - cmake: add option to enable/disable rtmp module - lock: use rwlock by default - cmake: fixes for MSVC 16 - json: fix win32 warnings - ci: add cmake build - mqueue: fix win32 warnings - tcp: fix win32 warnings - cmake: fix `target_link_libraries` for win32 - stun: fix win32 warnings - udp: fix win32 warnings - tls: fix win32 warnings - remove `HAVE_INTTYPES_H` - udp: fix win32 warnings - cmake: minor fixes - cmake: fix MSVC ninja - tcp: fix win32 warnings - udp: fix win32 msvc warnings - rtmp: fix win32 warning - bfcp: fix win32 warning - tls: fix libressl 3.5 - fix coverity scan warnings - Allow hanging up call that has not been ACKed yet - mk,cmake: add backtrace support and fix linking on OpenBSD - github: add CMake and Windows workflow - Windows (VS 2022/Ninja) - cmake: fixes for Android - tmr: reuse `tmr_jiffies_usec` - trace: use `gettid` as `thread_id` on linux - tmr: use `CLOCK_MONOTONIC_RAW` if defined - add atomic support - Sonarcloud - sip: fix gcc 6.3.0 warning for logical expression - add transport-cc rtcp feedback support # librem v2.0.0 (2022-03-12) - Restored rgb565 pixel format - vid: remove pixel formats RGB555 and RGB565 - cmake: version 3.7 - mk: bump dev version - au,aulevel: add `AUFMT_S32LE` - aubuf: add `aufbuf_resize()` - cmake: add `HAVE_UNISTD_H` check - cmake: add relative re include dir - cmake: minor fixes - mk: remove win32 project files - cmake: use version 3.10 - aubuf: fix `mem_deref` data race with `frame_destructor ` - h265: move packetizer from avcodec to rem - vidmix: fix `source_put` data race - vidmix: fix possible data race - h265: move `h265_is_keyframe` to rem - h265: move from avcodec to rem - preliminary support for CMake - gitignore: add vim swap and ctags files - ci: fix ccheck main repo path - aubuf: insert audio frames sorted by timestamp - auframe: add `auframe_update` - h264: fix win32 compiler cast warning - mk: bump version v1.0.0-dev3 - Increased debian compatibility level from 9 to 10 - aubuf: remove `aubuf_sort_auframe` return comment - aubuf: add `aubuf_sort_auframe()` - mk: cleanup cache directory - clangd: add config (headers only) - git: ignore clangd files - Fix win32 - mk: bump dev version - aubuf: add auframe functions - add resampler 16<->8 and 32<->16 kHz - aumix: add `aumix_source_mute` - update gitignore for visual studio artifacts - update PlatformToolset to vs2019 - mk: replace `pkg-config` modversion - mk: improve dependency - mk: ignore dependency check on `make clean` - debian: add `pkg- config` file - ci: remove ubuntu-16.04 test - mk: support more locations for `libre.pc` - mk: add `librem.pc` Makefile dependency - mk: add libre version check and pre-release - au/fmt: add `AUFMT_RAW` - auframe: use `enum aufmt` for format - auframe: move from baresip - h264: add functions from baresip - debian: fixes soname pkg build - mk: add abi versioning -------------------------------------------------------------------------------- ChangeLog: * Sun Mar 13 2022 Robert Scheck <robert@xxxxxxxxxxxxxxxxx> 2.0.0-1 - Upgrade to 2.0.0 (#2063450) * Thu Jan 20 2022 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 1.0.0-3 - Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild * Thu Jul 22 2021 Fedora Release Engineering <releng@xxxxxxxxxxxxxxxxx> - 1.0.0-2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild -------------------------------------------------------------------------------- References: [ 1 ] Bug #2019879 - [abrt] baresip: gtk_label_set_text(): baresip killed by SIGSEGV https://bugzilla.redhat.com/show_bug.cgi?id=2019879 [ 2 ] Bug #2063340 - libre-2.1.1 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063340 [ 3 ] Bug #2063450 - librem-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063450 [ 4 ] Bug #2063451 - baresip-2.0.0 is available https://bugzilla.redhat.com/show_bug.cgi?id=2063451 -------------------------------------------------------------------------------- ================================================================================ zabbix40-4.0.39-1.el7 (FEDORA-EPEL-2022-bd2c412d62) Open-source monitoring solution for your IT infrastructure -------------------------------------------------------------------------------- Update Information: Security fix for CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919 -------------------------------------------------------------------------------- ChangeLog: * Sat Mar 12 2022 Orion Poplawski <orion@xxxxxxxx> - 4.0.39-1 - Update to 4.0.39 -------------------------------------------------------------------------------- References: [ 1 ] Bug #2063280 - CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919 zabbix40: zabbix: Multiple security vulnerabilities [epel-all] https://bugzilla.redhat.com/show_bug.cgi?id=2063280 -------------------------------------------------------------------------------- ================================================================================ zabbix50-5.0.21-1.el7 (FEDORA-EPEL-2022-54fdcd70bd) Open-source monitoring solution for your IT infrastructure -------------------------------------------------------------------------------- Update Information: Security fix for CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919 -------------------------------------------------------------------------------- ChangeLog: * Sat Mar 12 2022 Orion Poplawski <orion@xxxxxxxx> - 5.0.21-1 - Update to 5.0.21 -------------------------------------------------------------------------------- References: [ 1 ] Bug #2063282 - CVE-2022-24349 CVE-2022-24917 CVE-2022-24918 CVE-2022-24919 zabbix50: zabbix: Multiple security vulnerabilities [epel-all] https://bugzilla.redhat.com/show_bug.cgi?id=2063282 -------------------------------------------------------------------------------- ================================================================================ zchunk-1.2.1-1.el7 (FEDORA-EPEL-2022-0d8982b43c) Compressed file format that allows easy deltas -------------------------------------------------------------------------------- Update Information: * Fix bug that prevented creating a zchunk file from a source that was larger than 2GB * Fix memory leak -------------------------------------------------------------------------------- ChangeLog: * Sat Mar 12 2022 Jonathan Dieter <jdieter@xxxxxxxxx> - 1.2.1-1 - Fixed bug that limited size of file that could be compressed using zchunk to 2GB - Fixed memory leak -------------------------------------------------------------------------------- _______________________________________________ epel-devel mailing list -- epel-devel@xxxxxxxxxxxxxxxxxxxxxxx To unsubscribe send an email to epel-devel-leave@xxxxxxxxxxxxxxxxxxxxxxx Fedora Code of Conduct: https://docs.fedoraproject.org/en-US/project/code-of-conduct/ List Guidelines: https://fedoraproject.org/wiki/Mailing_list_guidelines List Archives: https://lists.fedoraproject.org/archives/list/epel-devel@xxxxxxxxxxxxxxxxxxxxxxx Do not reply to spam on the list, report it: https://pagure.io/fedora-infrastructure
